Re: [Asterisk-Users] .:. .: .. .Stottering audio ??
Hi, Checked on the playbacks/voicecalls - only the playbacks have this problem (I am running Redhat - latest kernel version 2.4.19) Error Messages (results in stottering audio) NOTICE[1184091440]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! NOTICE[1184091440]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! | NOTICE[1184091440]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Anybody an idea how to solve this? Cheers Jeroen As far as I remember it is only in playbacks - voicecalls ok. Videocard in text mode? (does it have a special reason? Need to check how to do so) ::... . :: Specs ..::.. . .:; Pentium 4 - 2.4 GHz - 533 FSB 512 MB RAM Soundcard - soundblaster PCI16 (chip ES1373) Video - nVidia Gforce4 - 64 MB Jeroen Richard Alexander wrote: Only audio playbacks or on voice calls too ? Make sure the video card is in a text mode. What spec is the machine ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk - linux - JVB Sent: Thursday, August 14, 2003 2:22 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] .:. .: .. .Stottering audio ?? Importance: High Installed Asterisk on Redhat 9.0 - and not channeled to PSTN/PLMN networks (no XP100 or special hardware) yet When I use * with a softphone (SIP) - Asterisk answers the call but voicemail or other playbacks are STOTTERING for the first 30 secs (approx.)It happens more often when I start Asterisk with -vvvgc (less in -c). Following message displayed (loads of them!) NOTICE[1184091440]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Does anyone know where the problem is - what am I doing wrong or is it a known bug? Thanks in advance! Jeroen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems to install X100P
I'm using kernel 2.4.17. If I try to modprobe zaptel, I receive: Using /lib/modules/2.4.17/misc/zaptel.o /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol remove_proc_entry_Rsmp_5c747b84 /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol proc_mkdir_Rsmp_b1c61eb3 /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol create_proc_entry_Rsmp_a39969af /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol alloc_skb_Rsmp_0077ce3d /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol ppp_input_error_Rsmp_a7109407 /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol request_module_Rsmp_27e4dc04 /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol __kfree_skb_Rsmp_617cb2d1 /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol ppp_output_wakeup_Rsmp_d121fe32 /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol skb_over_panic_Rsmp_207f0a84 /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol ppp_unit_number_Rsmp_aab514cf /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol printk_Rsmp_1b7d4074 /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol softnet_data_Rsmp_6de03885 /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol ppp_register_channel_Rsmp_37546bfd /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol ppp_input_Rsmp_5330487b /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol ppp_unregister_channel_Rsmp_65ad2d68 /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol ppp_channel_index_Rsmp_390d5f4b What should I enable in my kernel to solve this? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems to install X100P
On Fri, 2003-08-15 at 17:55, [EMAIL PROTECTED] wrote: > I'm using kernel 2.4.17. > > If I try to modprobe zaptel, I receive: > > Using /lib/modules/2.4.17/misc/zaptel.o > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > remove_proc_entry_Rsmp_5c747b84 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > proc_mkdir_Rsmp_b1c61eb3 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > create_proc_entry_Rsmp_a39969af > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > alloc_skb_Rsmp_0077ce3d > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_input_error_Rsmp_a7109407 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > request_module_Rsmp_27e4dc04 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > __kfree_skb_Rsmp_617cb2d1 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_output_wakeup_Rsmp_d121fe32 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > skb_over_panic_Rsmp_207f0a84 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_unit_number_Rsmp_aab514cf > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol printk_Rsmp_1b7d4074 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > softnet_data_Rsmp_6de03885 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_register_channel_Rsmp_37546bfd > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_input_Rsmp_5330487b > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_unregister_channel_Rsmp_65ad2d68 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_channel_index_Rsmp_390d5f4b > > > What should I enable in my kernel to solve this? The last few certainly point to ppp missing. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems to install X100P
> > > > > > What should I enable in my kernel to solve this? > > The last few certainly point to ppp missing. On this machine I'm using PPP to connect to my ADSL and to connect to the pptp server in the company which I work for. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to detect process 2 frames
What does this error message mean? WARNING[262160]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 2 frames I've been getting these a lot lately, sound quality seems to have suffered. I'm using I4L driver with Fritz PCI ISDN card. However, even the regular echo test sounds a bit choppy over LAN connection. Happens both on CVS and 0.4.0 versions TIA, Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems to install X100P
On Fri, 2003-08-15 at 18:11, [EMAIL PROTECTED] wrote: > On this machine I'm using PPP to connect to my ADSL and to > connect to the pptp server in the company which I work for. Versions? Here it's kernel 2.4.20 and ppp-2.4.0-2 (only because of a problem with pppoatm.so) .config is:- CONFIG_PPP=m CONFIG_PPP_MULTILINK=y CONFIG_PPP_FILTER=y CONFIG_PPP_ASYNC=m CONFIG_PPP_SYNC_TTY=m CONFIG_PPP_DEFLATE=m CONFIG_PPP_BSDCOMP=m # CONFIG_PPPOE is not set CONFIG_PPPOATM=m # CONFIG_SLIP is not set Hope that helps -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Andy Powell wrote: Can't find the message in a search.. but below is a msg retreved from my archive.. this is what Mark sent a little while ago I have no idea if it actually does anything to the card, but on a modprobe I do get a msg saying it's using CTR21 Andy I'm in Paris right now and can't test this change, but I've been researching the DAA and there are a few international settings I can change, so I've changed the driver in CVS so that you can specify the operational mode. Try "modprobe wcfxo opermode=1" if you're in most of Europe and that should switch to CTR21 mode which slightly modifies a few of the electrical characteristics of the DAA. As we add modes you'll be able to see them with "modprobe wcfxo opermode=-1" and then doing a dmesg. Anyway all you folks that had some trouble like this try it out and let me know if it makes any difference. Mark I guess in his haste to help out the people who were having a problem, Mark looked at the wrong data sheet when he wrote this patch. I have corresponded with him and confirmed that this code requires the Global DAA chipset, which is not fitted to the current X100P cards. The mail I was referring to is: http://lists.digium.com/pipermail/asterisk-users/2003-August/017825.html Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
On Fri, 2003-08-15 at 11:42, Richard Scobie wrote: > I guess in his haste to help out the people who were having a problem, > Mark looked at the wrong data sheet when he wrote this patch. > > I have corresponded with him and confirmed that this code requires the > Global DAA chipset, which is not fitted to the current X100P cards. > So the question is does changing to CTR21 do anything at all with the existing cards? If it does nothing then OK, all I want to know is that it doesn't do anything bad. Much as I, and others, hate France Telecom I'd rather not nuke their exchange. I've got CID no problem. I have a problem with it not hanging up, but I am sure this is software not hardware, as it only happens if voicemail starts and the caller hangs up to exit, if they use # everything is OK. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk H323 Trunk
During debugging of H323 trunk side (using Jeremy Macnamara's H323
driver in ~/channels/h323) a couple of things come don't quite work
as advertised...
1/ the following line in extensions.conf explicitly sets the
outgoing caller ID (required in my case for downstream GK
processing..)
exten => _1NX.,1,SetCallerID,6400047602100
exten => _1NX.,2,Dial,H323/${EXTEN:1}
what comes out in the h323 datastream and logs is:-
dialled digits
{
{src digits}
{ "6400047602100)" }
...
note the unbalanced closing backet. We tried changing the
number length, and doing called ID with different functions,
but it looks like a bug.
A fairly detailed squiz around the digium site did not point
where to file a bug to, so I apologise for polluting this list..
=
When the h323 channel driver registers..
vipe50#sho gatek endp
GATEKEEPER ENDPOINT REGISTRATION
CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags
--- - --- - - -
202.37.19.101720 202.37.19.101024 zone1 TERM
E164-ID: 6400047602201
202.37.19.111720 202.37.19.111719 zone1 TERM
E164-ID: 6400047602999
;
; Asterisk, strongly preferred as a VOIP-GW, but registers as a TERM.
;
202.37.19.121720 202.37.19.1232829 zone1 TERM
H323-ID: fxchange
E164-ID: 6400047602100
;
; Cisco Call Manager, registering as a VOIP GW, with H323 Trunk
;
202.37.83.101720 202.37.83.101710 zone1 VOIP-GW
H323-ID: 202.37.83.10
203.79.85.252 1720 203.79.85.252 1719 zone1 TERM
E164-ID: 6400047601020
I had a look through the source, no comments stood out, any know a way
to get * registered as a VOIP-GW, rather than a TERM? Played with all
the obvious things in h323.conf
Many Thanks for reading this far...
Rgds Roger De Salis
--
\_Roger De Salis rdesalis at fx dot net dot nz
___
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Re: [Asterisk-Users] How can I know if a user is busy or not connected?
Hi John, It is not technically possible or it is not yet implemented? Thanks, Dan - Original Message - From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, August 15, 2003 1:05 AM Subject: Re: [Asterisk-Users] How can I know if a user is busy or not connected? > At 11:28 AM +0300 8/14/03, Dan wrote: > > > >Hi, > > > >I have defined several extensions. > >For some of them, the phone can be disconnected for a period of time (mobile > >users). > >When a call is initiated to that extension, if the user is not connected at > >that moment in time the caller see this as a busy extension. > >How can I detect from the extensions.conf file if a user is busy or it is > >not connected? > >I want to execute different applications in those two situations. > > > >Thanks, > >Dan > > You can't. > > Yes, that's very annoying. > > JT > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Transfer
Blind and assisted transfer work with Cisco 7960 phones. Blind transfer works fine with Budgetones. As long as you register to Asterisk. Jamie Carl wrote: Ok, just been thinking about this and thought I would ask before trying it out again. What is the state of SIP transfers? By this I mean transfers initiated via SIP messages, not via DTMF and '#'. Last time I tried, on X-Lite, clicking the transfer button dropped the call. Also, are/will both REFER and BYE/also methods be supported? To me, the SIP way of transfering is alot nicer and it seems silly to me to have a transfer button on your SIP phone that u can't use. Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '#' doesn't work for me
Do you have transfer turn on in zapata.conf? transfer=yes Hi, I cannot use '#' to initiate transfers. I have tried on different phones (7960, ATA, X-Lite). When I press '#' during a call, nothing happen. I have both T and t switches in Dial application. The transfer function works with Flash key on ATA, but in a very strange wayThe final destination is hunged up and then automatically called by the initial caller... This behavior request to put on hook the phone connected to the ATA in order to accept the transfer. During this period the phone is busy for the caller, so I must use some tricks in the dialing macro in order to acomodate this. Any other suggestions to better solve the transfer function? BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems to install X100P
Email class 101: If you want to start a new thread, START A NEW THREAD. Do not just change the subject line and remove the old text. Start a new message. It should be a law, there probably is a RFC for it. Kernel class 201: If your kernel is compiled with module versions (line noise looking stuff at the end of function names) then you need to have the proper modversion.h(?) file from when your kernel was compiled. With out it, nothing will work properly. Second, being 4 revisions behind head is not a good thing. Upgrade to 2.4.21 and preferably to one you compile yourself so you have all the right files to make your modules work right. Don't accept the crap you get from RH. On Fri, 2003-08-15 at 10:55, [EMAIL PROTECTED] wrote: > I'm using kernel 2.4.17. > > If I try to modprobe zaptel, I receive: > > Using /lib/modules/2.4.17/misc/zaptel.o > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > remove_proc_entry_Rsmp_5c747b84 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > proc_mkdir_Rsmp_b1c61eb3 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > create_proc_entry_Rsmp_a39969af > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > alloc_skb_Rsmp_0077ce3d > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_input_error_Rsmp_a7109407 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > request_module_Rsmp_27e4dc04 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > __kfree_skb_Rsmp_617cb2d1 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_output_wakeup_Rsmp_d121fe32 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > skb_over_panic_Rsmp_207f0a84 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_unit_number_Rsmp_aab514cf > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol printk_Rsmp_1b7d4074 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > softnet_data_Rsmp_6de03885 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_register_channel_Rsmp_37546bfd > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_input_Rsmp_5330487b > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_unregister_channel_Rsmp_65ad2d68 > /lib/modules/2.4.17/misc/zaptel.o: unresolved symbol > ppp_channel_index_Rsmp_390d5f4b > > > What should I enable in my kernel to solve this? > > Thanks, > > Isamar > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF SIP
Hello list, my case is as follows: SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the keypad on the phone. As suggested by you, I need to configure the SIP1 with out band dtmf mode , what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238 ? do I also need to make same kind configure for the SIP2 as well or I can configfure SIP2 with inband DTMF in sip.conf ? Thanks, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] '#' doesn't work for me
Hi, The problem was not related to ZAP channel. Even between two local SIP phone was the same. I have solve it just changing something in the ATA configuration and then butting back the old value. I still don't know the cause. Thanks, Dan - Original Message - From: "James Sizemore" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, August 15, 2003 3:34 PM Subject: Re: [Asterisk-Users] '#' doesn't work for me > Do you have transfer turn on in zapata.conf? > transfer=yes > > > >Hi, > > > >I cannot use '#' to initiate transfers. > >I have tried on different phones (7960, ATA, X-Lite). > >When I press '#' during a call, nothing happen. > >I have both T and t switches in Dial application. > >The transfer function works with Flash key on ATA, but in a very strange > >wayThe final destination is hunged up and then automatically called by > >the initial caller... This behavior request to put on hook the phone > >connected to the ATA in order to accept the transfer. During this period the > >phone is busy for the caller, so I must use some tricks in the dialing macro > >in order to acomodate this. > > > >Any other suggestions to better solve the transfer function? > > > > > >BR, > >Dan > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF SIP
Use rfc2833 for dtmf in your sip.conf.. 729 can't pass dtmf inband very well if at all. bkw On Fri, 15 Aug 2003, George Lin wrote: > > Hello list, > > my case is as follows: > > SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. > When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the > keypad on the phone. > > As suggested by you, I need to configure the SIP1 with out band dtmf mode , > what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238 > ? do I also need to make same kind configure for the SIP2 as well or I can > configfure SIP2 with inband DTMF in sip.conf ? > > Thanks, > > George Lin > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Autodialer / bulk dialer application
Hi all- I have asterisk running on 2 systems, with four E1 spans each. Each system is connected to a (big) NT DMS-100 switch. For load testing an IVR system running on one of the asterisk systems, I'd like to use the other system to generate a lot of outbound calls under program control - on most or all of its channels simultaneously. All of the asterisk dialplan and agi programming that I've done relate to handling inbound calls only. Is there a simple way to create a dialplan that will cause a line to make an outbound call of a specified duration, and then hangup? I can see how to do this in response to an incoming call on another line, but I just want asterisk to make outgoing test calls automatically when it starts up, on a bunch of channels over and over, until I kill it. Has anyone figured out how to do this? Thanks, Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I runAsterisk remotely from telnet session?
I am having problems trying to run asterisk from a telnet session. I am able to su to root and the command asterisk does not work. Any ideas why this may be occurring? I thought Asterisk could be configured remotely as well as run remotely? Thanks in advance… Steve Lane
Re: [Asterisk-Users] Asterisk H323 Trunk
H.323 doesn't have an explicit caller*id feature, so any callerid
specific features that have been added are hacks. Since you are using a
gatekeeper why don't you use a type=h323 to specify your H.323 id properly?
[6400047602100]
type=h323
secret=securepassword ; optional
Find me on IRC (JerJer) if u want to discuss this.
Jeremy McNamara
Roger De Salis wrote:
During debugging of H323 trunk side (using Jeremy Macnamara's H323
driver in ~/channels/h323) a couple of things come don't quite work
as advertised...
1/ the following line in extensions.conf explicitly sets the
outgoing caller ID (required in my case for downstream GK
processing..)
exten => _1NX.,1,SetCallerID,6400047602100
exten => _1NX.,2,Dial,H323/${EXTEN:1}
what comes out in the h323 datastream and logs is:-
dialled digits
{
{src digits}
{ "6400047602100)" }
...
note the unbalanced closing backet. We tried changing the
number length, and doing called ID with different functions,
but it looks like a bug.
A fairly detailed squiz around the digium site did not point
where to file a bug to, so I apologise for polluting this list..
=
When the h323 channel driver registers..
vipe50#sho gatek endp
GATEKEEPER ENDPOINT REGISTRATION
CallSignalAddr Port RASSignalAddr Port Zone Name Type
Flags
--- - --- - -
-
202.37.19.101720 202.37.19.101024 zone1 TERM
E164-ID: 6400047602201
202.37.19.111720 202.37.19.111719 zone1 TERM
E164-ID: 6400047602999
;
; Asterisk, strongly preferred as a VOIP-GW, but registers as a TERM.
;
202.37.19.121720 202.37.19.1232829 zone1 TERM
H323-ID: fxchange
E164-ID: 6400047602100
;
; Cisco Call Manager, registering as a VOIP GW, with H323 Trunk
;
202.37.83.101720 202.37.83.101710 zone1 VOIP-GW
H323-ID: 202.37.83.10
203.79.85.252 1720 203.79.85.252 1719 zone1 TERM
E164-ID: 6400047601020
I had a look through the source, no comments stood out, any know a way
to get * registered as a VOIP-GW, rather than a TERM? Played with all
the obvious things in h323.conf
Many Thanks for reading this far...
Rgds Roger De Salis
___
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Re: [Asterisk-Users] Can I runAsterisk remotely from telnet session?
Please explain, what do you mean by "does not work". It looks like a path problem. Instead of using "su" use "su -". "su" does not initialize the environment for the user you are "su"ing as, but "su -" will. /sbin and /usr/sbin a special path's that are only in root's environment. Hope this help, I will recommend reading a book on UNIX, this is UNIX 101 stuff. On Fri, 15 Aug 2003, Steve Lane wrote: > I am having problems trying to run asterisk from a telnet session. I am > able to su to root and the command asterisk does not work. Any ideas why > this may be occurring? I thought Asterisk could be configured remotely > as well as run remotely? > Thanks in advance. > > Steve Lane > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the highest quality codec I can usefor recording voice messages?
On 14/8/03 12:53, "WipeOut ." <[EMAIL PROTECTED]> wrote: > The highest quality codec is ulaw or alaw (otherwise know as G.711).. These > are the same as what comes in on your PSTN line.. > > If you want high Quality voice prompts your best bet os to record them on a PC > with a good quality mic and then copy then to your server.. You juast have to > make sure thay are recorded at the correct sample rate ( IIRC its 8khz 16bit > mono but I could be wrong on that).. > > Later.. > >> >> I have looked at the codec's available but I don't know how get the highest >> quality recorded message. >> >> If a user calls in over the normal telephone network is this limited to the >> carriers codec or the codec at the asterisk side? >> >> Would I get a higher quality result using VoIP rather than the normal >> network? >> >> Any help would be appreciated. >> >> Thanks >> Fats. >> I was interested in getting the highest quality over a normal phone line because I want users to be able to record their messages at the highest quality. They do it not me hence my question about the highest quality codec. If for example they used VoIP software on a computer could they get higher quality than over a normal phone line? If they did use a normal phone line can I increase the quality of is this a limitation at the carriers end? Any help would be appreciated. Thanks. Fats. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can I runAsterisk remotely from telnet session?
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > > Please explain, what do you mean by "does not work". It looks like a path > problem. Instead of using "su" use "su -". > > "su" does not initialize the environment for the user you are "su"ing as, > but "su -" will. /sbin and /usr/sbin a special path's that are only in > root's environment. Another thing I'm doing while soak testing an application (pre /etc/init.d startup script) is to run 'screen' as an unpriviledged user, then 'su -' to root (or even better, 'sudo su -') followed by starting the application. At that point CTRL-A CTRL-D will disconnect the session. Later, a quick 'screen -r' gets you back to that console. But agreed, a *NIX 101 book would be beneficial. HTH too, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?
> I was interested in getting the highest quality over a normal phone line > because I want users to be able to record their messages at the highest > quality. They do it not me hence my question about the highest quality > codec. > > If for example they used VoIP software on a computer could they get higher > quality than over a normal phone line? > > If they did use a normal phone line can I increase the quality of is this a > limitation at the carriers end? > > Any help would be appreciated. > > Thanks. > Fats. > Using alaw or ulaw over VoIP will give you best quality you are going to get over VoIP.. If your PSTN carrier is giving a bad quality line then you should probably change carrier... :) Most prompts will be converted to GSM files anyway so that will be the quality of the end result.. I wouldn't worry about it too much, just record some and take a listen, they should be fine.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can I runAsterisk remotely from telnet session?
Forgive me as my Linux and UNIX skills have not been utilized in over 6 months due to my job situation. I thank you for the suggestion of the "-" after the su command. It worked and I am slowly regaining my memory. Again I am sorry to ask such elementary questions, but the simple info helps me quite a bit when I leave out simple details. Highest regards, Steve Lane -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adams, Gavin Sent: Friday, August 15, 2003 12:43 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Can I runAsterisk remotely from telnet session? > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > > Please explain, what do you mean by "does not work". It looks like a path > problem. Instead of using "su" use "su -". > > "su" does not initialize the environment for the user you are "su"ing as, > but "su -" will. /sbin and /usr/sbin a special path's that are only in > root's environment. Another thing I'm doing while soak testing an application (pre /etc/init.d startup script) is to run 'screen' as an unpriviledged user, then 'su -' to root (or even better, 'sudo su -') followed by starting the application. At that point CTRL-A CTRL-D will disconnect the session. Later, a quick 'screen -r' gets you back to that console. But agreed, a *NIX 101 book would be beneficial. HTH too, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the highest quality codec I can usefor recording voice messages?
On 15/8/03 18:57, "WipeOut ." <[EMAIL PROTECTED]> wrote: >> I was interested in getting the highest quality over a normal phone line >> because I want users to be able to record their messages at the highest >> quality. They do it not me hence my question about the highest quality >> codec. >> >> If for example they used VoIP software on a computer could they get higher >> quality than over a normal phone line? >> >> If they did use a normal phone line can I increase the quality of is this a >> limitation at the carriers end? >> >> Any help would be appreciated. >> >> Thanks. >> Fats. >> > > Using alaw or ulaw over VoIP will give you best quality you are going to get > over VoIP.. If your PSTN carrier is giving a bad quality line then you should > probably change carrier... :) > > Most prompts will be converted to GSM files anyway so that will be the quality > of the end result.. > > I wouldn't worry about it too much, just record some and take a listen, they > should be fine.. > > Later.. Thanks for the fast reply. I assume you mean that alaw or ulaw is what the carrier delivers. Because my customers use the phones they have I do not have control over the carrier they use, I just meant it to mean all carriers. As in what is the standard format they deliver in. I assume that I have no control over how they deliver but at the Asterisk end I wanted to ensure the highest quality recording. If I used a direct connection from one computer to asterisk could I increase the quality by using a different codec, assuming they have broadband access at their end. I have read that the g711 is the highest quality in the sample rate but was not sure if this would still work with Asterisk or if I should use G729 as it seems to be double the bandwidth sampling at between 8 and 12 kps whereas g711 seems to sample at 64kbs but that may flood the connection. Any ideas? Thanks Fats. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I runAsterisk remotely from telnet session?
Personally I'd use ssh rather than telnet Andy *** REPLY SEPARATOR *** On 15/08/2003 at 12:21 Steve Lane wrote: >I am having problems trying to run asterisk from a telnet session. I am >able to su to root and the command asterisk does not work. Any ideas why >this may be occurring? I thought Asterisk could be configured remotely >as well as run remotely? >Thanks in advance. > >Steve Lane > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?
> Thanks for the fast reply. > > I assume you mean that alaw or ulaw is what the carrier delivers. Because my > customers use the phones they have I do not have control over the carrier > they use, I just meant it to mean all carriers. As in what is the standard > format they deliver in. > > I assume that I have no control over how they deliver but at the Asterisk > end I wanted to ensure the highest quality recording. > > If I used a direct connection from one computer to asterisk could I increase > the quality by using a different codec, assuming they have broadband access > at their end. I have read that the g711 is the highest quality in the sample > rate but was not sure if this would still work with Asterisk or if I should > use G729 as it seems to be double the bandwidth sampling at between 8 and 12 > kps whereas g711 seems to sample at 64kbs but that may flood the connection. > > Any ideas? > > Thanks > Fats. > Basically G.711 applies the least amout of compression and therefore the least amout of quality loss.. similarly G729 applies a very high level of compression so in theory would have a high quality loss.. Unfortuantely its not always that simple as not all codecs are created equal.. :) And as you said the G.711 having the highest quality also has the highest bandwidth requirement of 64Kbps (this increases to just over 80Kbps after you add IP overhead) so if you can sustain that leavel of throughput then the use of G.711 is irrelevant.. Like I said if you are really concerned then the best way is to test it and then listen to the results and see if you are happy with it.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I runAsterisk remotely from telnet session?
Repeat after me. Telnet BAD ssh good! bkw On Fri, 15 Aug 2003, Andy Powell wrote: > > Personally I'd use ssh rather than telnet > > Andy > > *** REPLY SEPARATOR *** > > On 15/08/2003 at 12:21 Steve Lane wrote: > > >I am having problems trying to run asterisk from a telnet session. I am > >able to su to root and the command asterisk does not work. Any ideas why > >this may be occurring? I thought Asterisk could be configured remotely > >as well as run remotely? > >Thanks in advance. > > > >Steve Lane > > > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
Dave Cotton wrote: So the question is does changing to CTR21 do anything at all with the existing cards? No as far as enabling CTR21 features, but the bits addressed in that register are labeled "Reserved", so to be correct the card should be left in the default FCC mode. If it does nothing then OK, all I want to know is that it doesn't do anything bad. Much as I, and others, hate France Telecom I'd rather not nuke their exchange. I've got CID no problem. I have a problem with it not hanging up, but I am sure this is software not hardware, as it only happens if voicemail starts and the caller hangs up to exit, if they use # everything is OK. Are you using busydetect=yes in zapata.conf? If not, try adding it and using BUSYDETECT_MARTIN and BUSYDETECT_COMPARE_TONE_AND_SILENCE when compiling asterisk. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the highest quality codec I can usefor recording voice messages?
On 15/8/03 20:15, "WipeOut ." <[EMAIL PROTECTED]> wrote: >> Thanks for the fast reply. >> >> I assume you mean that alaw or ulaw is what the carrier delivers. Because my >> customers use the phones they have I do not have control over the carrier >> they use, I just meant it to mean all carriers. As in what is the standard >> format they deliver in. >> >> I assume that I have no control over how they deliver but at the Asterisk >> end I wanted to ensure the highest quality recording. >> >> If I used a direct connection from one computer to asterisk could I increase >> the quality by using a different codec, assuming they have broadband access >> at their end. I have read that the g711 is the highest quality in the sample >> rate but was not sure if this would still work with Asterisk or if I should >> use G729 as it seems to be double the bandwidth sampling at between 8 and 12 >> kps whereas g711 seems to sample at 64kbs but that may flood the connection. >> >> Any ideas? >> >> Thanks >> Fats. >> > > Basically G.711 applies the least amout of compression and therefore the least > amout of quality loss.. similarly G729 applies a very high level of > compression so in theory would have a high quality loss.. > > Unfortuantely its not always that simple as not all codecs are created equal.. > :) > > And as you said the G.711 having the highest quality also has the highest > bandwidth requirement of 64Kbps (this increases to just over 80Kbps after you > add IP overhead) so if you can sustain that leavel of throughput then the use > of G.711 is irrelevant.. > > Like I said if you are really concerned then the best way is to test it and > then listen to the results and see if you are happy with it.. > > Later.. Yes you probably right. Test and see. So how do I get and use each codec to test them. I understand some are under tight copyright control. Do they have testing variations rather than buying first, find it it does not work, and your stuck with the cost? Thanks for you help. As always a fast reply. Thanks Fats. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the highest quality codec I can use for recording voice messages?
> > Yes you probably right. > Test and see. > > So how do I get and use each codec to test them. I understand some are under > tight copyright control. Do they have testing variations rather than buying > first, find it it does not work, and your stuck with the cost? > > > Thanks for you help. > > As always a fast reply. > Thanks > > Fats. > If you want to test under Asterisk using a SIP client I would suggest you download X-Lite.. This will give you G.711(64Kbps) and GSM(16Kbps) codecs that are both usable with Asterisk.. That should give you a basis for your comparison.. If you want to try out G.729 you will have to buy a licence for it from Digium and get a SIP phone that supports it.. Have fun.. Doubt you will get another fast reply cos its 2200 so bed will be calling soon.. :) Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO/FXS hotline
I was thinking of a hotline set up something like this: FXO --*--IAX--*--FXS The dialtone has to be provided by the remote end and flash hook has to be transparent Anyone have experience with hotlines on *? Would this work? cheers, darren --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.509 / Virus Database: 306 - Release Date: 8/12/03 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, quicknet and codecs (G729, 7231 Question)
Hi I am using asterisk with a Quicknet lineJack card. I am trying to get a proof of concept demo together before real deployment. I have a couple of qustions. 1) Can I use the codecs that are on Quicknet card (G.723, etc, etc). I tested it and I cannot use any codec in hardware, the only codecs that work are G711 and GSM. Since the lineJack provides G.723 in hardware, it should be available (other oh323 software can use it) 2) What would it take to get G729 for my demo. I cannot seem to find any link for this. I was told that I can use G729 as a paid option. I am willing to pay for it, but I do not know who to pay. Is trying to get 729 and 7231 working on asterisk worth it. Regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension and phone management bestpractices??
> > I guess my big question is: is it possible to have extensions mapped to > > people, not to phones? > > Yes, you just need to link the user/extension to a technology/channel > when logged in, and to a bogus value when not so that your dial will > fail quickly and fall through to voicemail. Also you will want to make > sure your voicemail goes out in email since you won't be able to get > stutter tone or MWI due to the channel not being assigned a extension. > -- > Steven Critchfield <[EMAIL PROTECTED]> Sorry I waited so long to reply to this.. I had to think about it a bit. Every reply I've had to this question has involved voicemail somehow. We don't use voicemail in the office - we have live operators handling all incoming calls, 24 hours a day. No voicemail. No IVRs. No auto attendant. So, another stupid question.. instead of failing over to voicemail, could it fail to a queue instead? - Devon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, quicknet and codecs (G729, 7231Question)
http://www.digium.com/index.php?menu=asterisk_g729 bwk On Fri, 15 Aug 2003 [EMAIL PROTECTED] wrote: > Hi > I am using asterisk with a Quicknet lineJack card. I am trying to get a > proof of concept demo together before real deployment. I have a couple of > qustions. > > 1) Can I use the codecs that are on Quicknet card (G.723, etc, etc). I > tested it and I cannot use any codec in hardware, the only codecs that > work are G711 and GSM. Since the lineJack provides G.723 in hardware, it > should be available (other oh323 software can use it) > > 2) What would it take to get G729 for my demo. I cannot seem to find any > link for this. I was told that I can use G729 as a paid option. I am > willing to pay for it, but I do not know who to pay. > > Is trying to get 729 and 7231 working on asterisk worth it. > > Regards > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registring soft phones in Asterisk
Can someone tell me where I can get information on how to configure the sip.conf file to register a soft phone? I tried the following entry and the phone will not register: [xten1] type=friend username=xxx secret=ppp host=dynamic I am using the X-Ten Lite soft phone, but I understood that I didn’t have the correct build. The next thing I did was download the SJLabs soft phone and tried it, and got the same thing. If I can get the Asterisk side (sip.conf) configured correctly, then I think configuring the soft phone will be a cakewalk. Can someone please help me with this? Thanks Steve Lane
Re: [Asterisk-Users] How can I know if a user is busy or notconnected?
Detecting what type of error or call result was produced by the "Dial" application has not yet been implemented, though it is desperately (IMHO) needed to allow the dialplan to more reasonably direct calls to the correct subsequent context. If I could code it, I would, but I can't, so I patiently wait... See my comments: http://lists.digium.com/pipermail/asterisk-users/2003-April/009797.html JT At 2:18 PM +0300 8/15/03, Dan wrote: Hi John, It is not technically possible or it is not yet implemented? Thanks, Dan - Original Message - From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, August 15, 2003 1:05 AM Subject: Re: [Asterisk-Users] How can I know if a user is busy or not connected? At 11:28 AM +0300 8/14/03, Dan wrote: > >Hi, > >I have defined several extensions. >For some of them, the phone can be disconnected for a period of time (mobile >users). >When a call is initiated to that extension, if the user is not connected at >that moment in time the caller see this as a busy extension. >How can I detect from the extensions.conf file if a user is busy or it is >not connected? >I want to execute different applications in those two situations. > >Thanks, >Dan You can't. Yes, that's very annoying. > JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: The Almighty X-Lite DTMF Problem (patch tested)
I am glad to hear this DTMF issue has been resolved. With this in mind I
would like to know if there is an index of Asterisk Consultants who
build/sell/support Asterisk IP PBXs in North America.
Xten on occasion receives sales inquiries for IP PBXs and I would like to
refer this business to the local Asterisk consultant.
Cheers,
Erik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jose
Ildefonso Camargo Tolosa
Sent: Thursday, August 14, 2003 9:39 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: The Almighty X-Lite DTMF Problem (patch
tested)
Hi!
I decided to apply Chris's patch for the rtp problem, it is working just
fine now. Thanks Chris!.
I think that Mark should submit it to the CVS.
Ildefonso.
[EMAIL PROTECTED]
>Pete,
>
>Try this patch below... I noticed that eStara's softphone has the same
>problem as xten's softphone when it comes to DTMF. Seems as though =
>Asterisk
>is not looking for the "end" bit per RFC2833. So try this fix. It =
>should
>do the trick (at least... it fixed mine).
>
>--Chris
>
>Index: rtp.c
>RCS file: /usr/cvsroot/asterisk/rtp.c,v
>retrieving revision 1.22
>diff -r1.22 rtp.c
>205a206
>>/ unsigned int event_end;
>/209a211,213
>>/ event_end = ntohl(*((unsigned int *)(data)));
>/>/ event_end <<= 8;
>/>/ event_end >>= 24;
>/224a229,234
>>/ else if(event_end & 0x80)
>/>/ {
>/>/ f = send_dtmf(rtp);
>/>/ resp = 0;
>/>/ }
>/>/
>/
>>/ -Original Message-
>/>/ From: [EMAIL PROTECTED]
[mailto:asterisk-dev-
>/>/ [EMAIL PROTECTED] ] On Behalf Of
[EMAIL PROTECTED]
>/>/ Sent: Tuesday, August 12, 2003 2:35 PM
>/>/ To: [EMAIL PROTECTED]
>/>/ Subject: [Asterisk-Dev] The Almighty X-Lite DTMF Problem
>/>/=20
>/>/ Hey guys,
>/>/=20
>/>/ I just was told by Rob at xten that the timestamp problem is fixed in =
>/the
>>/ rfc2833 implementation. I'm still having the exact same problems with
>/>/ voicemail(2) that I was before. Can someone please un-resolve bug 14 =
>/and
>>/ maybe I can work with someone to help debug what's happening?
>/>/=20
>/>/ Chris H, if you're still following this topic, fire me off an e-mail =
>/if
>>/ you want to see new debugs..
>/>/=20
>/>/ Thanks,
>/>/ Pete (km-)
>/>/ ___
>/>/ Asterisk-Dev mailing list
>/>/ [EMAIL PROTECTED]
>/>/ http://lists.digium.com/mailman/listinfo/asterisk-dev
>/
>
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension and phone management bestpractices??
On Fri, 2003-08-15 at 17:40, Devon Henderson wrote: > > > I guess my big question is: is it possible to have extensions mapped to > > > people, not to phones? > > > > Yes, you just need to link the user/extension to a technology/channel > > when logged in, and to a bogus value when not so that your dial will > > fail quickly and fall through to voicemail. Also you will want to make > > sure your voicemail goes out in email since you won't be able to get > > stutter tone or MWI due to the channel not being assigned a extension. > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > Sorry I waited so long to reply to this.. I had to think about it a bit. > > Every reply I've had to this question has involved voicemail somehow. We > don't use voicemail in the office - we have live operators handling all > incoming calls, 24 hours a day. No voicemail. No IVRs. No auto attendant. > > So, another stupid question.. instead of failing over to voicemail, could it > fail to a queue instead? Yes it can. Essentially instead of doing the failover to voicemail you will instruct it to goto the queue head end. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can I runAsterisk remotely from telnetsession?
On Fri, 2003-08-15 at 12:42, Adams, Gavin wrote: > > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > > > > Please explain, what do you mean by "does not work". It looks like a > path > > problem. Instead of using "su" use "su -". > > > > "su" does not initialize the environment for the user you are "su"ing > as, > > but "su -" will. /sbin and /usr/sbin a special path's that are only in > > root's environment. > > Another thing I'm doing while soak testing an application (pre > /etc/init.d startup script) is to run 'screen' as an unpriviledged user, > then 'su -' to root (or even better, 'sudo su -') followed by starting > the application. "sudo su -" is kind of a stange thing to do. You would probably be better of doing "sudo bash" as it also will give you a bash prompt with root login. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, quicknet and codecs (G729, 7231 Question)
Quicknet does not offer G.729. Make sure that you use the nixj driver for the quicknet card. It's better than the old driver shipped with the kernel. As soon as the new driver is done, we plan to send it to the kernel folks so it can replace the old one. I believe that You can get a 729 codec from digium, but I don't know how to make it work. [EMAIL PROTECTED] wrote: Hi I am using asterisk with a Quicknet lineJack card. I am trying to get a proof of concept demo together before real deployment. I have a couple of qustions. 1) Can I use the codecs that are on Quicknet card (G.723, etc, etc). I tested it and I cannot use any codec in hardware, the only codecs that work are G711 and GSM. Since the lineJack provides G.723 in hardware, it should be available (other oh323 software can use it) 2) What would it take to get G729 for my demo. I cannot seem to find any link for this. I was told that I can use G729 as a paid option. I am willing to pay for it, but I do not know who to pay. Is trying to get 729 and 7231 working on asterisk worth it. Regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I know if a user is busy or not connected?
You're right. Just some practical examples which really neeeds this feature. - using ATA, when the phone is off-hook with dialtone, if someone call that line it "sounds" busy, even the call waiting is activated. - when a phone is not registered, iy appears to be busy, which is not the right interpretation. You must handle in a different way those situations than in a busy one. BR, Dan - Original Message - From: "John Todd" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, August 15, 2003 11:54 PM Subject: Re: [Asterisk-Users] How can I know if a user is busy or not connected? > Detecting what type of error or call result was produced by the > "Dial" application has not yet been implemented, though it is > desperately (IMHO) needed to allow the dialplan to more reasonably > direct calls to the correct subsequent context. If I could code it, > I would, but I can't, so I patiently wait... > > See my comments: > http://lists.digium.com/pipermail/asterisk-users/2003-April/009797.html > > JT > > > At 2:18 PM +0300 8/15/03, Dan wrote: > > > >Hi John, > > > >It is not technically possible or it is not yet implemented? > > > >Thanks, > >Dan > > > >- Original Message - > >From: "John Todd" <[EMAIL PROTECTED]> > >To: <[EMAIL PROTECTED]> > >Sent: Friday, August 15, 2003 1:05 AM > >Subject: Re: [Asterisk-Users] How can I know if a user is busy or not > >connected? > > > > > >> At 11:28 AM +0300 8/14/03, Dan wrote: > >> > > >> >Hi, > >> > > >> >I have defined several extensions. > >> >For some of them, the phone can be disconnected for a period of time > >(mobile > >> >users). > >> >When a call is initiated to that extension, if the user is not connected > >at > >> >that moment in time the caller see this as a busy extension. > >> >How can I detect from the extensions.conf file if a user is busy or it is > >> >not connected? > >> >I want to execute different applications in those two situations. > >> > > >> >Thanks, > >> >Dan > >> > >> You can't. > >> > >> Yes, that's very annoying. > >> > > > JT > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
